[OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6 - SOLVED

[Chris Stone]

Well, looks like it WAS the ip_nat_sip and related kernel modules, but
not just on the Opensips server, also on the Asterisk server. I
unloaded all of the modules on the backend Asterisk server too and
tried a test call again and this time it worked just fine.

Appreciate all the help with this Henk and Ovidiu!



Thanks


Chris